3603results about How to "Improve sound quality" patented technology

A system and method for providing thermal gender recognition of people in any place of entertainment, leisure and / or recreation. The system utilizes infrared thermal imaging cameras and enables a prospective attendee determine in advance from a remote location where the prospective attendee would like to be. The system includes at least one computer system capable of communicating with a server over a network for retrieval, processing, and transmission of captured image data of each person in view of the infrared thermal imaging cameras. A website is configured to receive the image data of each person from the at least one computer system and implement the image data in a suitable format for viewing of the icons in various locations of the place by the prospective attendee from the remote location.

A hybrid audio encoding technique incorporates both ABR, or CBR, and VBR encoding modes. For each audio coding block, after a VBR quantization loop meets the NMR target, a second quantization loop might be called to adaptively control the final bitrate. That is, if the NMR-based quantization loop results in a bitrate that is not within a specified range, then a bitrate-based CBR or ABR quantization loop determines a final bitrate that is within the range and is adaptively determined based on the encoding difficulty of the audio data. Excessive bitrates from use of conventional VBR mode are eliminated, while still providing much more constant perceptual sound quality than use of conventional CBR mode can achieve.

A pit (52) is formed in a substrate comprising a silicon wafer (51) on a surface of which a silicon oxide thin layer (53) is formed. A sandwich structure (60) comprising a piezoelectric layer (62) and lower and upper electrodes (61, 63) joined to both surfaces of the piezoelectric layer is disposed so as to stride over the pit (52). The upper surface of the lower electrode (61) and the lower surface of the piezoelectric layer (62) joined to the upper surface of the lower electrode are treated so that the RMS variation of the height thereof is equal to 25 nm or less. The thickness of the lower electrode (61) is set to 150 nm or less. According to such a structure, there is provided a high-performance thin film bulk acoustic resonator which are excellent in electromechanical coupling coefficient and acoustic quality factor.

Patient may operate a CPAP system to deliver appropriate airway pressure home. A patient's apnea problem can be diagnosed at home without supervision with a CPAP device which delivers a continuously minimum appropriate pressure for substantially the entire period of therapy.

According to the first aspect of the invention, a hearing instrument with at least one microphone and signal processing comprises at least two receivers having a different frequency response. At least a first one of the receivers is placed outside the ear canal, for example in a behind-the-ear component. According to the second aspect of the invention, a hearing instrument comprising a behind-the-ear component and an external component for being placed in the user's ear canal or in the user's ear as well as a connection link between the behind-the-ear component and the external component is provided, where the connection link is reversibly pluggable to the behind-the-ear component and / or the in-the-ear-canal component and has a length that is reversibly adjustable. The hearing instrument also comprises fixation means for reversibly fixing the adjusted length of the connection link. A hearing instrument according to the third aspect comprises a fixation means separate from the in-the-ear-canal component. The fixation means may be positioned in the ear canal and rest therein. The in-the-ear-canal component may be connected to the fixation means and detached therefrom when the same is already in the ear canal. A hearing instrument according to the fourth aspect of the invention comprises an in-the-ear-canal component to be placed in a user's ear canal and fixation means for fixing it in the ear canal. The fixation means comprise an outer shell which is shaped to fit (i.e. custom shaped to fit the specific user's ear geometry) in the user's ear canal and a mounting structure for holding the in-the-ear-canal component. The outer shell is preferably resilient, i.e. has an elasticity allowing temporal deformation.

A system and method transmit a data stream from a source to a destination over a communication channel. A transmitter includes devices for processing inputs to assemble data packets for the data stream, and a multiplexer for assembling a data frame to be transmitted over the communication channel, in which each data frame has at least one fixed slot. The multiplexer sets at least one freely allocatable time slot in each data frame. Retransmission control devices connected to the multiplexer retransmit a specific data packet which is not properly received by the destination, using one of the freely allocatable slots.

A system for use with an audio reproduction system that corrects for distortion caused by the system as well as any hearing impairment suffered a listener. Test signals are introduced into the input of the system to produce test sounds that are perceptible by the listener. Using a pushbutton, the listener indicates when a test signal of progressively increasing volume reaches an audible level. The resulting measured values of the listener's threshold of hearing at different frequencies is compared with comparable data for a normal listener to generate correction values that are used to program an equalizer which compensates for not only the listener's hearing impairments but also any distortion produced by system components or room acoustics.

An audio decoding apparatus decodes high frequency component signals using a band expander that generates multiple high frequency subband signals from low frequency subband signals divided into multiple subbands and transmitted high frequency encoded information. The apparatus is provided with an aliasing detector and an aliasing remover. The aliasing detector detects the degree of occurrence of aliasing components in the multiple high frequency subband signals generated by the band expander. The aliasing remover suppresses aliasing components in the high frequency subband signals by adjusting the gain used to generate the high frequency subband signals. Thus occurrence of aliasing can be suppressed and the resulting degradation in sound quality can be reduced, even when real-valued subband signals are used in order to reduce the number of operations.

The present invention relates to a solution for improving the perceived sound quality of a decoded acoustic signal. The improvement is accomplished by means of extending the spectrum of a received narrow-band acoustic signal (aNB). According to the invention, a wide-band acoustic signal (aWB) is produced by extracting at least one essential attribute (zNB) from the narrow-band acoustic signal (aNB). Parameters, e.g. representing signal energies, with respect to wide-band frequency components outside the spectrum (ANB) of the narrow-band acoustic signal (aNB) are estimated based on the at least one essential attribute (zNB). This estimation involves allocating a parameter value to a wide-band frequency component, based on a corresponding confidence level. For instance, a relatively high parameter value is allowed to be allocated to a frequency component if it has a comparatively high degree certainty. In contrast, a relatively low parameter value is only allowed to be allocated to a frequency component if it is associated with a comparatively low degree certainty.

An energy corrector (105) for correcting a target energy for high-frequency components and a corrective coefficient calculator (106) for calculating an energy corrective coefficient from low-frequency subband signals are newly provided. These processors perform a process for correcting a target energy that is required when a band expanding process is performed on a real number only. Thus, a real subband combining filter and a real band expander which require a smaller amount of calculations can be used instead of a complex subband combining filter and a complex band expander, while maintaining a high sound-quality level, and the required amount of calculations and the apparatus scale can be reduced.

The present invention relates to a display unit for a refrigerator for improving sound quality. A display having a TV receiving function is installed at a front surface of a refrigerator door. A mounting panel with a plurality of speaker holes formed therein is also installed around an edge of the display. A speaker is mounted to a rear surface of the mounting panel on a portion where the speaker holes are formed. Further, a resonance box is installed to enclose the speaker such that low and middle frequency sound can be transmitted forward to the outside. A sound transmission tube that communicates with the speaker hole is formed at a front portion of the resonance box. In addition, a sound absorbing material is covered on an inner rear surface of the resonance box that faces the rear of the speaker.

A scalable audio coding / decoding method and apparatus are provided. The coding method includes the steps of (a) signal-processing input audio signals and quantizing the same for each predetermined coding band; (b) coding the quantized data corresponding to the base layer within a predetermined layer size; (c) coding the quantized data corresponding to the next enhancement layer of the coded base layer and the remaining quantized data uncoded and belonging to the enhancement layer, within a predetermined layer size; and (d) sequentially performing the layer coding steps for all layers, wherein the steps (b), (c) and (d) each comprise the steps of: (i) obtaining gamut bit allocation information representing the number of bits of the quantized data corresponding to the respective subbands belonging to a layer to be coded; (ii) obtaining the number of bits allocated to the respective subbands within each subband size of the layers; (iii) generating an index representing the presence of quantized data for predetermined frequency components forming the subbands for the quantized data corresponding to the number of allocated bits; and (iv) generating bitstreams by coding the quantized data corresponding to the gamut bit allocation information, quantization step size, index and number of bits allocated to the respective subbands, by a predetermined coding method.

A process for combining audio channels combines the audio channels to produce a combined audio channel and dynamically applies one or more of time, phase, and amplitude or power adjustments to the channels, to the combined channel, or to both the channels and the combined channel. One or more of the adjustments are controlled at least in part by a measure of auditory events in one or more of the channels and / or the combined channel. Applications include the presentation of multichannel audio in cinemas and vehicles. Not only methods, but also corresponding computer program implementations and apparatus implementations are included.

The invention provides a Chinese speech synthesis method based on phonemes and rhythm structures. The method is divided into a training stage and a synthesis stage, and comprises the steps of extracting therhythm structure features from the rhythm labeling information in a to-be-processed text according to the linguistic knowledge, and training a rhythm model based on the rhythm structure features; preprocessing the to-be-processed text and an audio to obtain a pinyin sequence containing the rhythm information and the corresponding acoustic features, then training an acoustic model, and deploying the trained rhythm model and acoustic model to a background; obtaining a text containing the rhythm information corresponding to the input text according to the rhythm model; converting into a pinyin sequence with rhythm information, and inputting into the acoustic model to obtain a linear frequency spectrum; and converting the linear frequency spectrum into the audio. According to the Chinesespeech synthesis method, the synthesized voice can be more natural, and especially for the longer clauses, the pause position can be determined. In addition, the model is deployed to the background at the synthesis stage, and the model loading time is saved, so that the voice synthesis speed can be increased.

Errors in pitch (frequency) allocation within a cochlear implant are corrected in order to provide a significant and profound improvement in the quality of sound perceived by the cochlear implant user. In one embodiment, the user is stimulated with a reference signal, e.g., the tone “A” (440 Hz) and then the user is stimulated with a probe signal, separated from the reference signal by an octave, e.g., high “A” (880 Hz). The user adjusts the location where the probe signal is applied, using current steering, until the pitch of the probe signal, as perceived by the user, matches the pitch of the reference signal, as perceived by the user. In this manner, the user maps frequencies to stimulation locations in order to tune his or her implant system to his or her unique cochlea.

Patients may operate a CPAP system to deliver appropriate airway pressure at their home. A patient's apnea problem ca be diagnosed at home without supervision with a CPAP device which delivers a continuously minimum appropriate pressure for substantially the entire period of therapy.